[asterisk-users] Reducing number of Asterisk processes?

Pezhman Lali lopl at lopl.net
Sat Jan 29 09:41:35 CST 2011


Dear
it's the default setting of asterisk.conf, your config is not complete.
-f in the ps output, shows your asterisk have been run in fork mode, disable
it.

[directories](!) ; remove the (!) to enable this
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk

[options]
;verbose = 3
;debug = 3
;alwaysfork = yes ; same as -F at startup
;nofork = yes ; same as -f at startup
;quiet = yes ; same as -q at startup
;timestamp = yes ; same as -T at startup
;execincludes = yes ; support #exec in config files
;console = yes ; Run as console (same as -c at startup)
;highpriority = yes ; Run realtime priority (same as -p at startup)
;initcrypto = yes ; Initialize crypto keys (same as -i at startup)
;nocolor = yes ; Disable console colors
;dontwarn = yes ; Disable some warnings
;dumpcore = yes ; Dump core on crash (same as -g at startup)
;languageprefix = yes ; Use the new sound prefix path syntax
;internal_timing = yes
;systemname = my_system_name ; prefix uniqueid with a system name for global
uniqueness issues
;autosystemname = yes ; automatically set systemname to hostname - uses
'localhost' on failure, or systemname if set
;maxcalls = 10 ; Maximum amount of calls allowed
;maxload = 0.9 ; Asterisk stops accepting new calls if the load average
exceed this limit
;maxfiles = 1000 ; Maximum amount of openfiles
;minmemfree = 1 ; in MBs, Asterisk stops accepting new calls if the amount
of free memory falls below this watermark
;cache_record_files = yes ; Cache recorded sound files to another directory
during recording
;record_cache_dir = /tmp ; Specify cache directory (used in conjunction with
cache_record_files)
;transmit_silence_during_record = yes ; Transmit SLINEAR silence while a
channel is being recorded
;transmit_silence = yes ; Transmit silence while a channel is in a waiting
state, a recording only state, or when DTMF is
                        ; being generated.  Note that the silence internally
is generated in raw signed linear format.
                        ; This means that it must be transcoded into the
native format of the channel before it can be sent
                        ; to the device.  It is for this reason that this is
optional, as it may result in requiring a
                        ; temporary codec translation path for a channel
that may not otherwise require one.
;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of
directly
;sendfullybooted = yes  ; Send the FullyBooted AMI event on AMI login and
when all modules are finished loading
;runuser = asterisk ; The user to run as
;rungroup = asterisk ; The group to run as
;lightbackground = yes ; If your terminal is set for a light-colored
background
documentation_language = en_US ; Set the Language you want Documentation
displayed in. Value is in the same format as locale names
;hideconnect = yes ; Hide messages displayed when a remote console connects
and disconnects

; Changing the following lines may compromise your security.
;[files]
;astctlpermissions = 0660
;astctlowner = root
;astctlgroup = apache
;astctl = asterisk.ctl

[compat]
pbx_realtime=1.6
res_agi=1.6
app_set=1.6

On Sat, Jan 29, 2011 at 4:32 PM, Gilles <codecomplete at free.fr> wrote:

> On Sat, 29 Jan 2011 15:47:53 +0330, Pezhman Lali <lopl at lopl.net>
> wrote:
> >check your /etc/asterisk/asterisk.conf  and post it here
>
> Here goes:
>
> root:/var/tmp> cat /etc/asterisk/asterisk.conf
> [directories]
> astetcdir => /etc/asterisk
> astmoddir => /usr/lib/asterisk/modules
> astvarlibdir => /var/lib/asterisk
> astagidir => /usr/share/asterisk/agi-bin
> astspooldir => /var/spool/asterisk
> astlogdir => /var/log/asterisk
>
> Thank you.
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110129/21673c96/attachment.htm>


More information about the asterisk-users mailing list